At present, packet-switched network has been widely used for its advantages, such as low price, flexible protocol, good scalability and mature technology. Many real-time services, such as IP telephony, TDM (Time Division Multiplexing) circuit emulation service and so on, have been or will be transferred over the packet-switched network. When real-time periodical service is being transferred over packet-switched network, due to queuing, congestion and variety in signal path, the network delay of the data packet keeps changing randomly, which is called network jitter delay. Thus, the time when the periodicity data packets sent from the sender arrive at the receiver is unsure, and it poses an obstacle for the receiver to resume the periodicity data packets sent from the sender. Therefore, how to absorb the network jitters introduced by packet-switched network is the key point for transferring periodical real-time services, such as TDM, in packet-switched network.
Currently, the common method for absorbing the network jitter is using a jitter buffer at the receiver. As shown in FIG. 1, data packets 1, 2, 3, 4, 5 and 6 from the sender 100 reach the receiver 120 over packet-switched network 110. Due to the existence of the delay jitter, the sequence of these packets can be disturbed when they pass the packet-switched network 110, and will be changed when they reach the receiver 120. The sequence of these packets is: 1, 3, 2, 5, 4 and 6. At the same time, the delay jitter brings about the network delay nd, and the receiver 120 uses jitter buffer 130 to absorb delay jitter, which causes the buffer delay bd. Thus, the gross output delay ted of data packets from the sender 100 to the jitter buffer is the sum of the network delay nd and the buffer delay bd.
In order to eliminate the delay jitter introduced by packet network, the size of the jitter buffer at least must be set to be 2 times packet rate×network jitter, and the normal working point of the jitter buffer at least must be set at packet rate×network jitter. For example, assuming the packet rate is 400 pks/s (packets/second), each packet has to experience 400×0.01=4 extra packets of queuing delays for absorbing 10 ms of network jitter. At present, there are mainly three types of methods for absorbing network jitters by using jitter buffers at the receiver. 1) A jitter buffer of fixed size. Such as the earlier experimental system for transferring TDM service on the Ethernet, in which the network jitter is assumed to be small, hence the fixed size jitter buffer is to be used. 2) A jitter buffer of successively increasing size. As an improvement to the first method, it uses the successively increasing size jitter buffer to absorb the largest network jitter. 3) Self-adaptive jitter buffer, which is a kind of jitter buffer whose size can be adjusted dynamically. This method has drawn growing attention in transferring TDM service over packet-switched network. This is mainly because of the wide range of services it undertakes, the complexity of the network, and the varied jitters under different circumstances of the network, in the practical packet-switched network. For example, a network jitter may be 20 ms at one period of time, 300 ms at the next period, and 10 ms in another next period. Obviously, neither the jitter buffer of fixed size nor the jitter buffer of successively increasing size fits the circumstances of the network, because the size of the jitter buffer will be set to absorb the 300 ms jitter (as for packets rate at 400 pks/s, the size of the jitter buffer is 240 packets) in both of the methods. Even if the network jitter decreases to 20 ms, the packets have to experience 120 extra packets queuing delays. Such a large delay is unsuitable for certain real-time applications. Therefore the jitter buffer must perform self-adaptive adjustment following the circumstances of the network. Under the above mentioned circumstances of the network, the packet delay is 120 packets when the circumstance is bad (300 ms jitter), however, it can be decreased to 4 packets when the circumstance turns better (10 ms) by using self-adaptive buffer. Generally, the traditional self-adaptive jitter buffer method adopts the prediction technique to the network jitter. The jitter prediction technique is based on either analyzing the jitters of the historically arrived packets, or directly on performing jitter prediction to changes of the filling level of the jitter buffer. For example, ajitter buffer adjustment method based on historically arrived packet jitter was presented in the article “An Empirical Study of a Jitter Management Scheme for Video Teleconferencing”, Donald L. Stone and Kevin Jeffay, Multimedia Systems Volume 2, Number 2, 1995; a self-adaptive jitter buffer adjustment method based on the changes of the filling level of the jitter buffer was presented in the article “An adaptive stream synchronization protocol”, written by Kurt Rothermel and Tobias Helbig and published in “Network and Operating System Support for Digital Audio and Video”, April 1995 pages 189-202, in which the architecture of self-adaptive adjustment theory is shown in FIG. 2. A large amount of threshold values and counters have been used in this self-adaptive adjustment procedure, HWM (High Water Mark) and LWM (Low Water Mark) are respectively defined as the high and low overflow threshold of the jitter buffer. Between HWM and LWM, and within the area of UTB (Upper Target Boundary) and LTB (Lower Target Boundary) is a target working zone. When the filling level is out of the working zone, the working parameters of the buffer will be adjusted by the self-adaptive procedure, and the filling level will be dragged back to the working zone, where HWM, LWM, UTB, LTB should all be adjusted according to the condition of network jitter. The self-adaptive adjustment method above was developed according to the characteristics of real-time services, and especially optimized for transferring real-time voice over the packet-switched network. More specifically, these methods should take into account the compromise between the end-to-end delay and the packet-dropping rate (as for real-time voice service, the typical value of the dropping rate is 5%). Generally speaking, the longer the buffer length is set to be, the larger the jitter can be absorbed, and the smaller the packets dropping rate is, but the longer the end-to-end delay is; on the contrary, the shorter the buffer length is set to be, the smaller the jitter can be absorbed, and the larger packets dropping rate is, but the shorter the end-to-end delay is. In practice, TDM service entails small packet-dropping rates, for example, the frame dropping rate of 2.048 Mbits/s E1 circuit simulation service on the Ethernet is defined to be lower than 7×10−6 by MEF (Metro Ethernet Forum). Obviously, if we take into account the packet dropping rate when adjusting the jitter buffer, quite a lot of statistic information is needed, and this will increase the complicacy and computational costs of self-adaptive adjustment algorithm.